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Allocate the memory with scoped/cleanup.h in audio_iio_aux_probe() to
reduce error handling (less error paths) and make the code a bit
simpler.
Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Link: https://patch.msgid.link/20240703-asoc-cleanup-h-v1-2-71219dfd0aef@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Allocate the memory with scoped/cleanup.h in audio_iio_aux_add_dapms()
to reduce error handling (less error paths) and make the code a bit
simpler.
Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Link: https://patch.msgid.link/20240703-asoc-cleanup-h-v1-1-71219dfd0aef@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
MIDI2 Set Tempo message defines the tempo in 10ns unit for finer
accuracy, while MIDI1 was defined in 1us unit. For adapting this
different unit, introduce "tempo_base" field to snd_seq_queue_tempo
struct so that user-space can pass the proper tempo base unit.
The accepted value is limited, it must be either 0, 10 or 1000.
The protocol version is bumped to 1.0.4 along with this.
The access with the older protocol version ignores the tempo-base
value in ioctls and always treats as 1000.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Link: https://patch.msgid.link/20240705160344.6481-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The internal mic boost on the VAIO models VJFE-CL and VJFE-IL is too high.
Fix this by applying the ALC269_FIXUP_LIMIT_INT_MIC_BOOST fixup to the machine
to limit the gain.
Signed-off-by: Edson Juliano Drosdeck <edson.drosdeck@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://patch.msgid.link/20240705141012.5368-1-edson.drosdeck@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Adding name-prefix for each audio controls is a redundant, because
name-prefix will be automatically added behind the control name when
creating a new control.
Signed-off-by: Shenghao Ding <shenghao-ding@ti.com>
Link: https://patch.msgid.link/20240705064846.1723-1-shenghao-ding@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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Merge v6.10-rc6 into drm-next
The exynos-next pull is based on a newer -rc than drm-next. hence
backmerge first to make sure the unrelated conflicts we accumulated
don't end up randomly in the exynos merge pull, but are separated out.
Conflicts are all benign: Adjacent changes in amdgpu and fbdev-dma
code, and cherry-pick conflict in xe.
Signed-off-by: Daniel Vetter <daniel.vetter@ffwll.ch>
Merge series from Herve Codina <herve.codina@bootlin.com>:
The qmc_audio driver supports only audio in interleaved mode.
Non-interleaved mode can be easily supported using several QMC channel
per DAI. In that case, data related to ch0 are sent to (received from)
the first QMC channel, data related to ch1 use the next QMC channel and
so on up to the last channel.
In terms of constraints and settings, the interleaved and
non-interleaved modes are slightly different.
In interleaved mode:
- The sample size should fit in the number of time-slots available for
the QMC channel.
- The number of audio channels should fit in the number of time-slots
(taking into account the sample size) available for the QMC channel.
In non-interleaved mode:
- The number of audio channels is the number of available QMC
channels.
- Each QMC channel should have the same number of time-slots.
- The sample size equals the number of time-slots of one QMC channel.
This series add support for the non-interleaved mode in the qmc_audio
driver and is composed of the following parts:
- Patches 1 and 2: Fix some issues in the qmc_audio
- Patches 3 to 6: Prepare qmc_audio for the non-interleaved mode
- Patches 7 and 8: Extend the QMC driver API
- Patches 9 and 10: The support for non-interleaved mode itself
Compared to the previous iteration, this v2 series mainly improves
qmc_audio_access_is_interleaved().
Merge series from Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>:
Improve a bit the Qualcomm LPASS RX macro driver and align similar parts
of code with LPASS WSA macro driver for consistency.
No external dependencies.
Set "Speaker Force Firmware Load" as the common kcontrol
for both tas27871 and tas2563 and move it into newly-created
tasdevice_snd_controls, and keep the digital gain and analog
gain in tas2781_snd_controls.
Signed-off-by: Shenghao Ding <shenghao-ding@ti.com>
Link: https://patch.msgid.link/20240704094939.1824-1-shenghao-ding@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Laptop 10431A63 contains valid _DSD, but missing Speaker ID
description. Add this discription, but keep the rest of the _DSD to
ensure the correct firmware and tuning is loaded for this laptop.
Signed-off-by: Stefan Binding <sbinding@opensource.cirrus.com>
Link: https://patch.msgid.link/20240703140802.27688-1-sbinding@opensource.cirrus.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
acpi_get_first_physical_node() can return NULL in several cases (no such
device, ACPI table error, reference count drop to 0, etc).
Existing check just emit error message, but doesn't perform return.
Then this NULL pointer is passed to devm_acpi_dev_add_driver_gpios()
where it is dereferenced.
Adjust this error handling by adding error code return.
Found by Linux Verification Center (linuxtesting.org) with SVACE.
Fixes: 02527c3f23 ("ASoC: amd: add Machine driver for Jadeite platform")
Signed-off-by: Aleksandr Mishin <amishin@t-argos.ru>
Link: https://patch.msgid.link/20240703191007.8524-1-amishin@t-argos.ru
Signed-off-by: Mark Brown <broonie@kernel.org>
Driver does not unregister typec structures (typec_mux_dev and
typec_switch_desc) during removal leading to leaks. Fix this by moving
typec registering parts to separate function and using devm interface to
release them. This also makes code a bit simpler:
- Smaller probe() function with less error paths and no #ifdefs,
- No need to store typec_mux_dev and typec_switch_desc in driver state
container structure.
Cc: stable@vger.kernel.org
Fixes: 10f514bd17 ("ASoC: codecs: Add WCD939x Codec driver")
Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Reviewed-by: Neil Armstrong <neil.armstrong@linaro.org>
Link: https://patch.msgid.link/20240701122616.414158-1-krzysztof.kozlowski@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
commit c721f189e8 ("reset: Instantiate reset GPIO controller for
shared reset-gpios") check if there is no "resets" property
will fallback to "reset-gpios".
So don't need to handle "reset-gpios" separately in the driver,
the "reset-gpios" handler is duplicated with "resets" control handler,
remove it.
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Link: https://patch.msgid.link/1720009575-11677-1-git-send-email-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
There is no need to keep separate token list for dai and 'common' copier
token list when the 'common' list is actually the aif list, the
SOF_COPIER_DEEP_BUFFER_TOKENS are not applicable for buffers.
We could have separate lists for all types but it is probably simpler to
just use a single list for all types of copiers. Function specific tokens
will be only parsed by function specific code anyways.
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://patch.msgid.link/20240704085944.371450-1-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The current fsl_qmc_audio works in interleaved mode. The audio samples
are interleaved and all data are sent to (received from) one QMC
channel.
Using several QMC channels, non interleaved mode can be easily
supported. In that case, data related to ch0 are sent to (received from)
the first QMC channel, data related to ch1 use the next QMC channel and
so on up to the last channel.
In terms of constraints and settings, the two modes are slightly
different:
- Interleaved mode:
- The sample size should fit in the number of time-slots available
for the QMC channel.
- The number of audio channels should fit in the number of
time-slots (taking into account the sample size) available for the
QMC channel.
- Non-interleaved mode:
- The number of audio channels is the number of available QMC
channels.
- Each QMC channel should have the same number of time-slots.
- The sample size equals the number of time-slots of one QMC
channel.
Add support for the non-interleaved mode allowing multiple QMC channel
per DAI. The DAI switches in non-interleaved mode when more that one QMC
channel is available.
Signed-off-by: Herve Codina <herve.codina@bootlin.com>
Link: https://patch.msgid.link/20240701113038.55144-11-herve.codina@bootlin.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Constraints are set by qmc_dai_startup(). These constraints are specific
to the interleaved mode.
With the future introduction of support for non-interleaved mode, a new
set of constraints will be set. To make the code clear and keep
qmc_dai_startup() simple, extract the current interleaved mode
constraints settings to a specific function.
Signed-off-by: Herve Codina <herve.codina@bootlin.com>
Link: https://patch.msgid.link/20240701113038.55144-7-herve.codina@bootlin.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Submitting data to QMC channels is done in several places: transfer
completions and DAI start. The operation done is simple and consist in
one function call.
With the future introduction of support for non-interleaved mode,
submitting data will be more complex.
To avoid copy/paste of code in several places, introduce
qmc_audio_pcm_{read,write}_submit() whose goal is to handle this
data submission.
Signed-off-by: Herve Codina <herve.codina@bootlin.com>
Link: https://patch.msgid.link/20240701113038.55144-6-herve.codina@bootlin.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The current QMC audio driver uses only one QMC channel per DAI. The
context used by QMC channel transfer (read and write) completion
routines does not contains any QMC channel and the only one available
per DAI is used to schedule the next transfer.
This works pretty well with only one QMC channel per DAI.
The future support for non-inlerleave mode will use several QMC channel
per DAI. In that case, QMC channel transfer completion routines need to
identify the QMC channel related to the completion.
In order to fill this lack, even if identifying the current QMC channel
among several QMC channels is not needed for the current code, add one
indirection level and introduce the qmc_dai_chan data structrure.
This structure contains the QMC channel involved in the completion and
refererences to the runtime context (capture and playback) used by the
DAI.
Signed-off-by: Herve Codina <herve.codina@bootlin.com>
Link: https://patch.msgid.link/20240701113038.55144-5-herve.codina@bootlin.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The driver mixes some internal values for channel DMA buffer handling
and PCM pointer handling. In the currently supported interleaved mode,
this mix does not lead to any issues but in order to prepare the
support for the non-interleaved mode, having them clearly separated will
ease the support and avoid additional computation to convert values used
in channel DMA buffer management in values usable for PCM pointer.
Use a specific set of variable for PCM pointer handling and an other set
for channel DMA buffer.
Signed-off-by: Herve Codina <herve.codina@bootlin.com>
Link: https://patch.msgid.link/20240701113038.55144-4-herve.codina@bootlin.com
Signed-off-by: Mark Brown <broonie@kernel.org>
./scripts/checkpatch.pl --strict --codespell detected several issues
when running on the fsl_qmc_audio.c file:
- CHECK: spaces preferred around that '*' (ctx:VxV)
- CHECK: Alignment should match open parenthesis
- CHECK: Comparison to NULL could be written "!prtd"
- CHECK: spaces preferred around that '/' (ctx:VxV)
- CHECK: Lines should not end with a '('
- CHECK: Please don't use multiple blank lines
Some of them are present several times.
Fix all of these issues without any functional changes.
Signed-off-by: Herve Codina <herve.codina@bootlin.com>
Link: https://patch.msgid.link/20240701113038.55144-3-herve.codina@bootlin.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The HDaudio specification Section 3.6.2 limits the number of BDL entries to 256.
Make sure we don't allow more periods than this normative value.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://patch.msgid.link/20240704090106.371497-1-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Those registers will be used when JD source is RT711_JD2_1P8V_1PORT.
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Acked-by: Shuming Fan <shumingf@realtek.com>
Link: https://patch.msgid.link/20240704092327.652609-1-yung-chuan.liao@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
When system enters suspend with an active stream, SOF core
calls hw_params_upon_resume(). On Intel platforms with HDA DMA used
to manage the link DMA, this leads to call chain of
hda_dsp_set_hw_params_upon_resume()
-> hda_dsp_dais_suspend()
-> hda_dai_suspend()
-> hda_ipc4_post_trigger()
A bug is hit in hda_dai_suspend() as hda_link_dma_cleanup() is run first,
which clears hext_stream->link_substream, and then hda_ipc4_post_trigger()
is called with a NULL snd_pcm_substream pointer.
Fixes: 2b009fa082 ("ASoC: SOF: Intel: hda: Unify DAI drv ops for IPC3 and IPC4")
Link: https://github.com/thesofproject/linux/issues/5080
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://patch.msgid.link/20240704085708.371414-1-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Driver's probe() has two allocations which are needed only within the
probe() itself - for devm_regmap_init_mmio().
Usage of devm interface is a bit misleading here, because these can be
freed right after each scope finishes.
This makes the code a bit more obvious and self documenting.
Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Link: https://patch.msgid.link/20240701-b4-qcom-audio-lpass-codec-cleanups-v3-6-6d98d4dd1ef5@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Driver uses ARRAY_SIZE() to get number of widgets later passed to
snd_soc_dapm_new_controls(), which is an 'unsigned int'.
Reviewed-by: Dmitry Baryshkov <dmitry.baryshkov@linaro.org>
Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Link: https://patch.msgid.link/20240701-b4-qcom-audio-lpass-codec-cleanups-v3-5-6d98d4dd1ef5@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Number of widgets in array passed to snd_soc_dapm_new_controls() cannot
be negative, so make it explicit by using 'unsigned int', just like
snd_soc_add_component_controls() is doing.
Reviewed-by: Dmitry Baryshkov <dmitry.baryshkov@linaro.org>
Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Link: https://patch.msgid.link/20240701-b4-qcom-audio-lpass-codec-cleanups-v3-4-6d98d4dd1ef5@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
The driver has static 'struct regmap_config', which is then customized
depending on device version. This works fine, because there should not
be two devices in a system simultaneously and even less likely that such
two devices would have different versions, thus different regmap config.
However code is cleaner and more obvious when static data in the driver
is also const - it serves as a template.
Mark the 'struct regmap_config' as const and duplicate it in the probe()
with kmemdup to allow customizing per detected device variant.
Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Link: https://patch.msgid.link/20240701-b4-qcom-audio-lpass-codec-cleanups-v3-3-6d98d4dd1ef5@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Switch the driver to use GPIO descriptors.
Notice that we let the gpiolib handle line inversion for the
active low reset line (nreset !reset).
There are no upstream device trees using the tas5086 compatible
string, if there were, we would need to ascertain that they all
set the GPIO_ACTIVE_LOW flag on their GPIO lines.
Signed-off-by: Linus Walleij <linus.walleij@linaro.org>
Link: https://patch.msgid.link/20240701-asoc-tas-gpios-v1-1-d69ec5d79939@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Merge series from Richard Fitzgerald <rf@opensource.cirrus.com>:
These two commits set the upper limit of the Speaker Volume control
to +12dB instead of +100dB.
This should have been a simple 1-line change to the #define in the
header file, but only the HDA cs35l56 driver is using this define.
The ASoC cs35l56 driver was using hardcoded numbers instead of the
header defines.
So the first commit changes the ASoC driver to use the #defined
constants. The second commit corrects the value of the constant.
'devmodel' hasn't actually been used since:
'commit 3275158fa5 ("parport: remove use of devmodel")'
and everyone now has it set to true and has been fixed up; remove
the flag.
(There are still comments all over about it)
Signed-off-by: Dr. David Alan Gilbert <linux@treblig.org>
Acked-by: Sudip Mukherjee <sudipm.mukherjee@gmail.com>
Link: https://lore.kernel.org/r/20240502154823.67235-4-linux@treblig.org
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
In the current flow all interrupts are disabled in runtime suspend
phase. However interrupts enablement only exists in fsl_xcvr_prepare().
After resume fsl_xcvr_prepare() may not be called so it will cause all
interrupts still disabled even if resume from suspend. Interrupts
should be explictily enabled after resume.
Also, DPATH reset setting only exists in fsl_xcvr_prepare(). After
resume from suspend DPATH should be reset otherwise there'll be channel
swap issue.
Signed-off-by: Chancel Liu <chancel.liu@nxp.com>
Acked-by: Shengjiu Wang <shengjiu.wang@gmail.com>
Link: https://patch.msgid.link/20240628094354.780720-1-chancel.liu@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The "Speaker Volume" control was being defined using four hardcoded magic
numbers. There are #defines in the cs35l56.h header for these numbers, so
change the code to use the defined constants.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://patch.msgid.link/20240703095517.208077-2-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Replace the open-coded parsing of "reg" with of_property_read_reg().
The #ifdef is also easily replaced with IS_ENABLED().
Signed-off-by: Rob Herring (Arm) <robh@kernel.org>
Link: https://patch.msgid.link/20240702215402.839673-1-robh@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Replace the open-coded parsing of "reg" with of_property_read_reg().
The #ifdef is also easily replaced with IS_ENABLED().
Signed-off-by: Rob Herring (Arm) <robh@kernel.org>
Link: https://patch.msgid.link/20240702215349.839350-1-robh@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Merge series from Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>:
Code used to create standalone and widget controls is mostly same, with
with exception that in standalone case dynamic object needs to be
registered and control created directly.
Following patches clean up and unify kcontrol creation code in topology
code.
Code used to create standalone and widget enum control is same, with
exception that in standalone case dynamic object needs to be registered
and control created directly.
Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Link: https://patch.msgid.link/20240627101850.2191513-14-amadeuszx.slawinski@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Code used to create standalone and widget mixer control is same, with
exception that in standalone case dynamic object needs to be registered
and control created directly.
Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Link: https://patch.msgid.link/20240627101850.2191513-13-amadeuszx.slawinski@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Code used to create standalone and widget bytes control is same, with
exception that in standalone case dynamic object needs to be registered
and control created directly.
Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Link: https://patch.msgid.link/20240627101850.2191513-12-amadeuszx.slawinski@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
soc_tplg_dbytes_create() missed se->dobj.index initialization, so add it
there. Additionally separate dynamic object initialization into separate
logical block code.
Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Link: https://patch.msgid.link/20240627101850.2191513-9-amadeuszx.slawinski@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Widget kcontrols do not need to be managed as topology dynamic objects
with an index and a linked list. As they are always associated with a
widget which is already a topology dynamic object, thus all
addition/removals of a widget will by design manage the kcontrol.
Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Link: https://patch.msgid.link/20240627101850.2191513-3-amadeuszx.slawinski@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Instead of using goto, when there is no controls, just do a loop when
there are. Overall the check seems to be a bit redundant as
num_kcontrols will only be above 0 if kcontrols are set anyway, but
let's keep it, while simplifying code.
Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Link: https://patch.msgid.link/20240627101850.2191513-2-amadeuszx.slawinski@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Merge series from Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>:
simple-audio-mux is designed to be used generally, thus "Input 1" or
"Input 2" are used to selecting MUX input. This numbered inputs would work,
but might be not user friendly in some case, for example in case of system
hardware design has some clear labels.
Adds new "state-labels" property and enable to select MUX by own state names.
Original
> amixer set "MUX" "Input 1"
> amixer set "MUX" "Input 2"
Use mux-names
sound_mux: mux {
compatible = "simple-audio-mux";
mux-gpios = <...>;
=> state-labels = "Label_A", "Label_B";
};
> amixer set "MUX" "Label_A"
> amixer set "MUX" "Label_B"
Merge series from srinivas.kandagatla@linaro.org:
Existing way of allocating soundwire master ports on Qualcommm platforms is
dynamic, and in linear order starting from 1 to MAX_PORTS.
This will work as long as soundwire device ports are 1:1 mapped
linearly. However on most Qcom SoCs like SM8550, SM8650, x1e80100, these
are NOT mapped in that order.
The result of this is that only one speaker among the pair of speakers
is always silent, With recent changes for WSA codec to support codec
versions and along with these patches we are able to get all speakers
working on these SoCs.
Optimize the memory usage in struct snd_pcm_runtime - use boolean
value for the standard sync ID scheme.
Introduce snd_pcm_set_sync_per_card function to build synchronization
IDs.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://patch.msgid.link/20240625172836.589380-3-perex@perex.cz
Until the commit e11f0f90a6 ("ALSA: pcm: remove SNDRV_PCM_IOCTL1_INFO
internal command"), there was a possibility to pass information
about the synchronized streams to the user space. The mentioned
commit removed blindly the appropriate code with an irrelevant comment.
The revert may be appropriate, but since this API was lost for several
years without any complains, it's time to improve it. The hardware
parameters may change the used stream clock source (e.g. USB hardware)
so move this synchronization ID to hw_params as read-only field.
It seems that pipewire can benefit from this API (disable adaptive
resampling for perfectly synchronized PCM streams) now.
Note that the contents of ID is not supposed to be used for direct
comparison with a specific byte sequence. The "empty" case is when
all bytes are zero (driver does not offer this information)
and all other cases must be only used for equal comparison among
PCM streams (including different sound cards) if they are using
identical hardware clock.
Cc: Takashi Sakamoto <takaswie@kernel.org>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://patch.msgid.link/20240625172836.589380-2-perex@perex.cz
The 'wsa->dev' is assigned closer to the end of the probe() function, so
the dev_err() must not use it - it is still NULL at this point. Instead
there is already a local 'dev' variable.
Fixes: 727de4fbc5 ("ASoC: codecs: lpass-wsa-macro: Correct support for newer v2.5 version")
Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Reviewed-by: Dmitry Baryshkov <dmitry.baryshkov@linaro.org>
Link: https://patch.msgid.link/20240628095831.207942-2-krzysztof.kozlowski@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
The 'rx->dev' is assigned closer to the end of the probe() function, so
the dev_err() must not use it - it is still NULL at this point. Instead
there is already a local 'dev' variable.
Fixes: dbacef0589 ("ASoC: codec: lpass-rx-macro: prepare driver to accomdate new codec versions")
Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Reviewed-by: Dmitry Baryshkov <dmitry.baryshkov@linaro.org>
Link: https://patch.msgid.link/20240628095831.207942-1-krzysztof.kozlowski@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
bitfield.h is not explicitly included but it is required for FIELD_PREP
to be expanded by the preprocessor. If it is not implicitly included,
there will be a compiler error (as seen with ARCH=hexagon allmodconfig):
sound/soc/fsl/lpc3xxx-i2s.c:169:10: error: call to undeclared function 'FIELD_PREP'; ISO C99 and later do not support implicit function declarations [-Wimplicit-function-declaration]
169 | tmp |= LPC3XXX_I2S_WW8 | LPC3XXX_I2S_WS_HP(LPC3XXX_I2S_WW8_HP);
| ^
sound/soc/fsl/lpc3xxx-i2s.h:42:30: note: expanded from macro 'LPC3XXX_I2S_WW8'
42 | #define LPC3XXX_I2S_WW8 FIELD_PREP(0x3, 0) /* Word width is 8bit */
| ^
sound/soc/fsl/lpc3xxx-i2s.c:205:34: error: call to undeclared function 'FIELD_PREP'; ISO C99 and later do not support implicit function declarations [-Wimplicit-function-declaration]
205 | LPC3XXX_I2S_DMA1_TX_EN | LPC3XXX_I2S_DMA0_TX_DEPTH(4));
| ^
sound/soc/fsl/lpc3xxx-i2s.h:65:38: note: expanded from macro 'LPC3XXX_I2S_DMA0_TX_DEPTH'
65 | #define LPC3XXX_I2S_DMA0_TX_DEPTH(s) FIELD_PREP(0xF0000, s) /* Set the DMA1 TX Request level */
| ^
sound/soc/fsl/lpc3xxx-i2s.c:210:34: error: call to undeclared function 'FIELD_PREP'; ISO C99 and later do not support implicit function declarations [-Wimplicit-function-declaration]
210 | LPC3XXX_I2S_DMA0_RX_EN | LPC3XXX_I2S_DMA1_RX_DEPTH(4));
| ^
sound/soc/fsl/lpc3xxx-i2s.h:70:38: note: expanded from macro 'LPC3XXX_I2S_DMA1_RX_DEPTH'
70 | #define LPC3XXX_I2S_DMA1_RX_DEPTH(s) FIELD_PREP(0x700, s) /* Set the DMA1 RX Request level */
| ^
Include bitfield.h explicitly, so that FIELD_PREP is always expanded,
clearing up the compiler error.
Fixes: 0959de657a ("ASoC: fsl: Add i2s and pcm drivers for LPC32xx CPUs")
Signed-off-by: Nathan Chancellor <nathan@kernel.org>
Link: https://patch.msgid.link/20240701-lpc32xx-asoc-fix-include-for-field_prep-v1-1-0c5d7f71921b@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
clang points out that ret may be used uninitialized in
lpc32xx_i2s_probe() in an error pointer path (which becomes fatal with
CONFIG_WERROR):
sound/soc/fsl/lpc3xxx-i2s.c:326:47: error: variable 'ret' is uninitialized when used here [-Werror,-Wuninitialized]
326 | "failed to init register map: %d\n", ret);
| ^~~
sound/soc/fsl/lpc3xxx-i2s.c:310:9: note: initialize the variable 'ret' to silence this warning
310 | int ret;
| ^
| = 0
1 error generated.
One solution would be a small refactoring of the second parameter in
dev_err_probe(), PTR_ERR(i2s_info_p->regs), to be the value of ret in
the if statement. However, a nicer solution for debugging purposes,
which is the point of this statement, would be to use the '%pe'
specifier to symbolically print the error pointer value. Do so, which
eliminates the uninitialized use of ret, clearing up the warning.
Fixes: 0959de657a ("ASoC: fsl: Add i2s and pcm drivers for LPC32xx CPUs")
Signed-off-by: Nathan Chancellor <nathan@kernel.org>
Link: https://patch.msgid.link/20240701-lpc32xx-asoc-fix-uninitialized-ret-v1-1-985d86189739@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Add name_prefix as the prefix name of DSP firmwares
and calibrated data files which stored speaker
calibrated impedance.
Signed-off-by: Shenghao Ding <shenghao-ding@ti.com>
Link: https://patch.msgid.link/20240629101112.628-1-shenghao-ding@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Add support to parse static master port map information from device tree.
This is required for correct port mapping between soundwire device and
master ports.
Reviewed-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Tested-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Reviewed-by: Neil Armstrong <neil.armstrong@linaro.org>
Tested-by: Neil Armstrong <neil.armstrong@linaro.org> # on SM8650-HDK
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-by: Dmitry Baryshkov <dmitry.baryshkov@linaro.org>
Link: https://patch.msgid.link/20240626-port-map-v2-4-6cc1c5608cdd@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Add support to parse static master port map information from device tree.
Reviewed-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Tested-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Tested-by: Neil Armstrong <neil.armstrong@linaro.org> # on SM8650-HDK
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-by: Dmitry Baryshkov <dmitry.baryshkov@linaro.org>
Link: https://patch.msgid.link/20240626-port-map-v2-2-6cc1c5608cdd@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
simple-audio-mux is designed to be used generally, thus "Input 1" or
"Input 2" are used to selecting MUX input. This numbered inputs would
work, but might be not user friendly in some case, for example in case
of system hardware design has some clear labels.
Adds new "state-labels" property and enable to select MUX by own state
names.
Original
> amixer set "MUX" "Input 1"
Use mux-names
sound_mux: mux {
compatible = "simple-audio-mux";
mux-gpios = <...>;
state-labels = "Label_A", "Label_B";
};
> amixer set "MUX" "Label_A"
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://patch.msgid.link/87msn27xpg.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Add compatible string and specific soc data to support rpmsg sound card
on i.MX95 platform.
Signed-off-by: Chancel Liu <chancel.liu@nxp.com>
Acked-by: Shengjiu Wang <shengjiu.wang@gmail.com>
Link: https://patch.msgid.link/20240626071202.7149-2-chancel.liu@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
struct sdw_slave_prop is zero-initialized by the SoundWire core so there
is no need to clear clk_stop_mode1 to false. Removing this also avoids
having an unnecessary build dependency on a struct member.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://patch.msgid.link/20240701104444.172556-4-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
No product was ever released with A1 silicon so there is no
need for the driver to include support for it.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://patch.msgid.link/20240701104444.172556-3-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch reverts a series of commits that allowed for the ASP
registers to be owned by either the driver or the firmware. Nothing
currently depends on the functionality that is being reverted, so
it is safe to remove.
The commits being reverted are (last 3 are bugfixes to the first 2):
commit 72a77d7631
("ASoC: cs35l56: Fix to ensure ASP1 registers match cache")
commit 07f7d6e7a1
("ASoC: cs35l56: Fix for initializing ASP1 mixer registers")
commit 4703b014f2
("ASoC: cs35l56: fix reversed if statement in cs35l56_dspwait_asp1tx_put()")
commit c14f09f010
("ASoC: cs35l56: Fix deadlock in ASP1 mixer register initialization")
commit dfd2ffb373
("ASoC: cs35l56: Prevent overwriting firmware ASP config")
These reverts have been squashed into a single commit because there
would be no reason to revert only some of them (which would just
reintroduce bugs).
The changes introduced by the commits were well-intentioned but
somewhat misguided. ACPI does not provide any information about how
audio hardware is linked together, so that information has to be
hardcoded into drivers. On Windows the firmware is customized to
statically setup appropriate configuration of the audio links,
and the intent of the commits was to re-use this information if the
Linux host drivers aren't taking control of the ASP. This would
avoid having to hardcode the ASP config into the machine driver on
some systems.
However, this added complexity and race conditions into the driver.
It also complicates implementation of new code.
The only case where the ASP is used but the host is not taking
ownership is when CS35L56 is used in SoundWire mode with the ASP
as a reference audio interconnect. But even in that case it's not
necessarily required even if the firmware initialized it. Typically
it is used to avoid the host SDCA drivers having to be capable of
aggregating capture paths from multiple SoundWire peripherals. But
the SOF SoundWire support is capable of doing that aggregation.
Reverting all these commits significantly simplifies the driver.
Let's just use the normal Linux mechanisms of the machine driver and
ALSA controls to set things up instead of trying to use the firmware
to do use-case setup.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://patch.msgid.link/20240701104444.172556-2-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Merge series from Piotr Wojtaszczyk <piotr.wojtaszczyk@timesys.com>:
This pach set is to bring back audio to machines with a LPC32XX CPU.
The legacy LPC32XX SoC used to have audio spport in linux 2.6.27.
The support was dropped due to lack of interest from mainaeners.
Introduce support for Cirrus Logic Device CS40L50: a
haptic driver with waveform memory, integrated DSP,
and closed-loop algorithms.
The ASoC driver enables I2S streaming to the device.
Reviewed-by: David Rhodes <drhodes@opensource.cirrus.com>
Signed-off-by: James Ogletree <jogletre@opensource.cirrus.com>
Reviewed-by: Jeff LaBundy <jeff@labundy.com>
Reviewed-by: Ricardo Rivera-Matos <rriveram@opensource.cirrus.com>
Reviewed-by: Mark Brown <broonie@kernel.org>
Link: https://lore.kernel.org/r/20240620161745.2312359-6-jogletre@opensource.cirrus.com
Signed-off-by: Lee Jones <lee@kernel.org>
clk_prepare_enable() may fail, so we should better check its return
value and propagate it in the case of error.
Fixes: 62a7fc32a6 ("ASoC: max98088: Add master clock handling")
Signed-off-by: Chen Ni <nichen@iscas.ac.cn>
Link: https://patch.msgid.link/20240628080534.843815-1-nichen@iscas.ac.cn
Signed-off-by: Mark Brown <broonie@kernel.org>
The dummy DAI should allow any (reasonable) rates possible.
Make the rate continuous for dummy and set range from 5512Hz to 768kHz
The change is mostly cosmetic as dummy is skipped when setting
the hwparams.
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Link: https://patch.msgid.link/20240628120130.2015665-1-jbrunet@baylibre.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The Qualcomm LPASS WSA macro codec driver uses now parts of common
module, so it has to select SND_SOC_LPASS_MACRO_COMMON.
sound/soc/codecs/lpass-wsa-macro.o: in function `wsa_macro_probe':
sound/soc/codecs/lpass-wsa-macro.c:2767:(.text+0x1c9c): undefined reference to `lpass_macro_get_codec_version'
Reported-by: kernel test robot <lkp@intel.com>
Closes: https://lore.kernel.org/oe-kbuild-all/202406272231.th1LtuLk-lkp@intel.com/
Fixes: 5dcf442bbb ("ASoC: codecs: lpass-wsa-macro: Prepare to accommodate new codec versions")
Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Link: https://patch.msgid.link/20240627125203.171048-1-krzysztof.kozlowski@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Merge series from Shengjiu Wang <shengjiu.wang@nxp.com>:
The transmitter and receiver part of the SAI interface need to be
configured with different master/slave mode, especially to work
with the audiomix module.
The SAI1 TX is in master mode, but SAI1 RX is in slave mode.
So add another two DAIs for TX and RX separately in fsl_sai driver.
There will be three devices for audiomix sound card, hw:x,0 is
the playback device for one SAI, hw:x,1 is the playback device
for another SAI, hw:x,2 is the capture device for audmix
output.
Merge series from Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>:
Introduce the ability for sound cards to manually order the startup of
the various components in the card.
A relatively large batch of updates, largely due to the long interval
since I last sent fixes due to various travel and holidays. There's a
lot of driver specific fixes and quirks in here, none of them too major,
and also some fixes for recently introduced memory safety issues in the
topology code.
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Merge tag 'asoc-fix-v6.10-rc5' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v6.10
A relatively large batch of updates, largely due to the long interval
since I last sent fixes due to various travel and holidays. There's a
lot of driver specific fixes and quirks in here, none of them too major,
and also some fixes for recently introduced memory safety issues in the
topology code.
The version B will support the multi-lane function and integrate the DMIC function
in one SoundWire interface.
Due to some registers having different default values between version A and B,
this patch also removes the redundant default registers to avoid confusion.
Signed-off-by: Shuming Fan <shumingf@realtek.com>
Link: https://patch.msgid.link/20240620103237.2124196-1-shumingf@realtek.com
Signed-off-by: Mark Brown <broonie@kernel.org>
If IORESOURCE_MEM "lpass-rxtx-cdc-dma-lpm" or "lpass-va-cdc-dma-lpm"
resources is not provided in Device Tree due to any error,
platform_get_resource_byname() will return NULL which is later
dereferenced. According to sound/qcom,lpass-cpu.yaml, these resources
are provided, but DT can be broken due to any error. In such cases driver
must be able to protect itself, since the DT is external data for the
driver.
Adjust this issues by adding NULL return check.
Found by Linux Verification Center (linuxtesting.org) with SVACE.
Fixes: b138706225 ("ASoC: qcom: Add regmap config support for codec dma driver")
Signed-off-by: Aleksandr Mishin <amishin@t-argos.ru>
Link: https://patch.msgid.link/20240605104953.12072-1-amishin@t-argos.ru
Signed-off-by: Mark Brown <broonie@kernel.org>
There will be three devices for this sound card, hw:x,0 is
the playback device for one SAI, hw:x,1 is the playback device
for another SAI, hw:x,2 is the capture device for audmix
output. then capture device and playback device can be configured
with different master/slave mode.
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Link: https://patch.msgid.link/1718174452-17596-4-git-send-email-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
As audmix requires playback and capture stream in different
master/slave mode, so separate playback and capture stream to
different DAI. There are three DAIs required, two DAIs for playback
one DAI for capture.
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Link: https://patch.msgid.link/1718174452-17596-3-git-send-email-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The transmitter and receiver part of the SAI interface need to be
configured with different master/slave mode, especially to work
with the audiomix module.
+-------+ +-----------+
| SAI1 | --TX--> | |
| | <--RX-- | |
+-------+ | |
| AUDIOMIX |
+-------+ | |
| SAI2 | --TX--> | |
+-------+ +-----------+
The SAI1 TX is in master mode, but SAI1 RX is in slave mode.
So add another two DAIs for TX and RX separately. but only
defined fsl_sai_set_dai_fmt_tx() and fsl_sai_set_dai_fmt_rx()
ops function for current case, in the future, the other ops
function for TX and RX can be defined if required.
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Link: https://patch.msgid.link/1718174452-17596-2-git-send-email-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
